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Author SHA1 Message Date
houssainy@google.com
5d3e7ac1a3 Check on the existence of report directory 
Reports will be written at rtcBot/test/reports/<report_name>

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7372 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-06 17:21:27 +00:00
pbos@webrtc.org
42684be21b Wire up CPU adaptation in WebRtcVideoEngine2. 
Includes clean-up work to be able to use the webrtc::Call::Config that's
set up. This introduced a CallFactory to spawn a FakeCall with the
config used and allowed removal of FakeWebRtcVideoChannel2.

BUG=1788
R=mflodman@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7370 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-03 11:25:45 +00:00
henrike@webrtc.org
31b75eae05 Moves xmllite's unittests to rtc_unittest. 
BUG=3836
R=andrew@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26669005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7369 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-02 18:43:47 +00:00
glaznev@webrtc.org
25cc745d6b Switch to SW video decoder on Android after getting 2 or more 
critical errors from HW decoder.

BUG=410730
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7368 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-02 16:58:05 +00:00
henrik.lundin@webrtc.org
4b133da5fd Let RtpFileSource use RtpFileReader 
RtpFileSource used to implement it's own RTP dump file reader, but
with this change it simply uses RtpFileReader. One benefit is that
pcap files are now also supported.

All NetEq test tools that use RtpFileSource are updated.

BUG=2692
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7367 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-02 08:19:38 +00:00
bjornv@webrtc.org
348eac641e audio_processing: Replaced WEBRTC_SPL_RSHIFT_U32 with >> 
A trivial macro that is replaced. Affected components:
* AGC
* NSx

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29599005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7366 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-02 08:07:05 +00:00
sergeyu@chromium.org
5fa8c458d8 Remove mouse cursor capturer from the ScreenCapturer interface 
Mouse can be captured using MouseCursorMonitor and all code in chromium
already uses it instead of ScreenCapturer.

R=jiayl@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7363

Review URL: https://webrtc-codereview.appspot.com/31529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7365 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-02 01:47:10 +00:00
sergeyu@chromium.org
6138f0f89d Revert "Remove mouse cursor capturer from the ScreenCapturer interface" 
This reverts commit 0adc4953512ee0a57cf7f3c0591b024c2316554a. It broke
FYI bots

TBR=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/27649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7364 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-02 01:36:43 +00:00
sergeyu@chromium.org
1fced0f2aa Remove mouse cursor capturer from the ScreenCapturer interface 
Mouse can be captured using MouseCursorMonitor and all code in chromium
already uses it instead of ScreenCapturer.

R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7363 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-02 00:18:10 +00:00
sergeyu@chromium.org
76819d315d Add error trap for XFixesGetCursorImage() 
BUG=https://code.google.com/p/webrtc/issues/detail?id=3245
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7362 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 23:07:12 +00:00
andrew@webrtc.org
325cff01b4 Import LappedTransform and friends. 
Add code for doing block-based frequency domain processing. Developed
and reviewed in isolation. Corresponding export CL:
https://chromereviews.googleplex.com/95187013/

R=bercic@google.com, kjellander@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7359 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 17:42:18 +00:00
henrike@webrtc.org
593c3a0868 rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7358 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 16:33:03 +00:00
henrike@webrtc.org
4530b2ca48 Revert 7355 "Fix parallelization in libjingle_p2p_unittest." 
Breaks waterfall.

TBR=pbos@webrtc.org
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/22909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7357 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 15:43:55 +00:00
henrike@webrtc.org
36b0c1afae Adds PRESUBMIT.py dispensation for depending on rtc_base. 
Dispensation for: a few test suites, desktop capture and libjingle.

BUG=N/A
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7356 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 14:40:58 +00:00
pbos@webrtc.org
fd29205e6e Fix parallelization in libjingle_p2p_unittest. 
Adding VirtualSocketServers to SessionTest and RelayServerTest to avoid
contention on real ports.

R=juberti@webrtc.org
BUG=2597
TEST=third_party/gtest-parallel/gtest-parallel -w 64 out/Debug/libjingle_p2p_unittest

Review URL: https://webrtc-codereview.appspot.com/26679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7355 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 12:31:31 +00:00
pbos@webrtc.org
c86e45d7c4 Fix parallelizability in modules_tests. 
R=henrik.lundin@webrtc.org
BUG=3873
TEST=third_party/gtest-parallel/gtest-parallel -r 10 -w 64 out/Debug/modules_tests

Review URL: https://webrtc-codereview.appspot.com/24799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7354 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 10:05:40 +00:00
henrik.lundin@webrtc.org
4cebd84c79 Reland "Remove DTMF status methods from Voice Engine" r7276 
This reverts r7277.

TBR=henrika@webrtc.org,pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7353 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 08:23:21 +00:00
kjellander@webrtc.org
4e4fe4f9ae Add support for MSan 
Add third_party/instrumented_libraries to setup_links.py
Add tools/msan/blacklist.txt which is the default location used
by MSan.

These changes are prerequisites to be able to use MSan with WebRTC.
To use it, one must also run:
sudo third_party/instrumented_libraries/install-build-deps.sh
to get the instrumented libraries installed (requires
/etc/apt/sources.list to be setup with deb-src entries).

NOTICE: Compilation is not yet working, but with this we can setup
a FYI bot to work with.

BUG=chromium:416871
TESTED=gclient sync + generate projects using:
GYP_DEFINES='msan=1 use_instrumented_libraries=1 instrumented_libraries_jobs=20' webrtc/build/gyp_webrtc
Built successfully in Release and ran a couple of tests (some crashed, some passed).

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7352 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 08:03:19 +00:00
kjellander@webrtc.org
afefed5c93 Update checkdeps.py rules in DEPS 
The initial rules didn't allow including
source from third_party, which is incorrect.
Cleanup irrelevant rules for directories that
are ignored, since WebRTC don't have any source
code in those locations.

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7351 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 06:03:47 +00:00
henrike@webrtc.org
83fe69da95 Added presubmit protecting against inclusion of rtc_base, while allowing rtc_base_approved. 
BUG=N/A
R=andrew@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7349 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 21:54:26 +00:00
kjellander@webrtc.org
3037bc3447 GN: Add common configs to tools and test. 
Similar changes as in https://review.webrtc.org/28589004/
were missed in https://review.webrtc.org/25569004/.
This should fix the Chromium WebRTC FYI bots that currently
are broken due to lack of include paths.

BUG=3441
TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7347 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 19:07:58 +00:00
kjellander@webrtc.org
b8caf6a504 GN: Enable libvpx, add link target and convert some test targets 
Libvpx now supports GN and this CL turns on compiling it.
I also introduced an executable target 'webrtc_tests'
that depends on all in WeBRTC + tests in order to get a full
linking step executed (since we've seen link problems for GN
when rolling WebRTC into Chromium).

I also converted a few test targets and made a GN file for
third_party/gflags.

BUG=3441
TESTED=Trybots + full Chromium build with a symlinked src/third_party/webrtc
dir to a workspace wit this CL applied.

R=brettw@chromium.org
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7344 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 18:05:02 +00:00
andrew@webrtc.org
d05756f0a2 Changed mips_arch_variant variable value corresponding to Chromium code changes. 
Chromium commit URL: https://crrev.com/c8a5da7455b57b2399e4a69e8100c098d9870052

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23809004

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7343 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 15:53:24 +00:00
xians@webrtc.org
79a7148108 Revert 7337 "Reland 28629004: adding new AEC dump start interfac..." 
> Reland 28629004: adding new AEC dump start interface for chrome
> 
> adding new AEC dump start interface for chrome.
> 
> This is required because we are not allow to pass CRT objects across dll boundaries, that says, when we pass a file descriptor from chrome dll to libpeerconnection dll, the file descriptor will become invalid immediate, more information can be found here:
> http://msdn.microsoft.com/en-us/library/ms235460.aspx
> 
> Chromium bug:crbug/415935
> TEST=bots
> R=bjornv@webrtc.org, kwiberg@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/27639004

TBR=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7342 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 15:29:13 +00:00
xians@webrtc.org
7aad5e5cce Revert 7338 "Fixed the android build by making the interface pur..." 
> Fixed the android build by making the interface pure virtual.
> 
> TBR=asapersson@webrtc.org, bjornv@webrtc.org,
> 
> Review URL: https://webrtc-codereview.appspot.com/24789004

TBR=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7341 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 15:26:15 +00:00
houssainy@google.com
d0bb5862f5 Collecting stats every fixed time in webrtc_video_streaming.js test 
and prepare the format these collected stats to be plotted using one of
external dev-tools.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7340 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 15:20:15 +00:00
andrew@webrtc.org
db75a66b0f Minor code change to fix some warnings in MIPS build. 
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26619004

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7339 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 15:17:50 +00:00
xians@webrtc.org
90d1979d77 Fixed the android build by making the interface pure virtual. 
TBR=asapersson@webrtc.org, bjornv@webrtc.org,

Review URL: https://webrtc-codereview.appspot.com/24789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7338 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 15:15:22 +00:00
xians@webrtc.org
14092e00f1 Reland 28629004: adding new AEC dump start interface for chrome 
adding new AEC dump start interface for chrome.

This is required because we are not allow to pass CRT objects across dll boundaries, that says, when we pass a file descriptor from chrome dll to libpeerconnection dll, the file descriptor will become invalid immediate, more information can be found here:
http://msdn.microsoft.com/en-us/library/ms235460.aspx

Chromium bug:crbug/415935
TEST=bots
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7337 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 14:35:15 +00:00
henrike@webrtc.org
792d1a0541 Adds isolate for rtc_unittests and moves sound's unittests to rtc_unittest. 
BUG=N/A
R=andrew@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7336 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 14:21:10 +00:00
xians@webrtc.org
875206196c Revert 7334 "adding new AEC dump start interface for chrome." 
> adding new AEC dump start interface for chrome.
> 
> This is required because we are not allow to pass CRT objects across dll boundaries, that says, when we pass a file descriptor from chrome dll to libpeerconnection dll, the file descriptor will become invalid immediate, more information can be found here:
> http://msdn.microsoft.com/en-us/library/ms235460.aspx
> 
> Chromium bug:crbug/415935
> TEST=bots
> R=bjornv@webrtc.org, kwiberg@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/28629004

TBR=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7335 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 13:30:05 +00:00
xians@webrtc.org
2e417d6428 adding new AEC dump start interface for chrome. 
This is required because we are not allow to pass CRT objects across dll boundaries, that says, when we pass a file descriptor from chrome dll to libpeerconnection dll, the file descriptor will become invalid immediate, more information can be found here:
http://msdn.microsoft.com/en-us/library/ms235460.aspx

Chromium bug:crbug/415935
TEST=bots
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7334 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 13:11:27 +00:00
henrik.lundin@webrtc.org
38c121c484 Minor modifications to test::RtpFileReader 
Adding original_length to the Packet struct. This is populated with
the plen value from the RTP dump file. In the case of reading a
pcap file, original_length will be equal to length.

Also increasing the maximum packet size to 3500 bytes. This is to
accomodate some test files that contain PCM16b audio encoding.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7333 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 11:08:44 +00:00
pbos@webrtc.org
1795c358fc Add default implementation of Add/RemoveObserver. 
Needed to remove Add/RemoveObserver from RTCVideoEncoderFactory in
Chromium before removing these completely. This is done to keep the
chromium.webrtc.fyi bots happy and to make rolling webrtc revisions
easier.

R=stefan@webrtc.org
BUG=3876

Review URL: https://webrtc-codereview.appspot.com/23839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7332 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 09:45:25 +00:00
bjornv@webrtc.org
65e56dba53 audio_processing/aecm: Added help function for calculating log of energy 
The same operation of calculating log of the energy was executed four times. This CL adds a help function LogOfEnergyInQ8() for that.

BUG=N/A
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7331 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 09:31:28 +00:00
bjornv@webrtc.org
23ec8372a6 audio_processing: Removed usage of macro WEBRTC_SPL_MUL 
WEBRTC_SPL_MUL is a trivial multiplication after casting to int32_t. This is already taken care of by the compiler which makes the macro unnecessary.

Affected components:
* AGC
* NSx

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7330 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 09:29:28 +00:00
bjornv@webrtc.org
750423c722 audio_processing: Replaced trivial macro WEBRTC_SPL_LSHIFT_W32 with << 
Affected components:
* AECM
* AGC
* HPF
* NSx

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7329 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 09:26:36 +00:00
kjellander@webrtc.org
8cad9432d5 Revert 7327 "Update isolate.gypi files + link to isolate_driver.py" 
Breaks debug compilation (didn't run all trybots when testing this).

> Update isolate.gypi files + link to isolate_driver.py
> 
> This updates the isolate.gypi copies we're forced to
> maintain in our code repo to Chromium revision c264a05.
> 
> Since isolated testing is now using a new launch script
> in tools: isolate_driver.py, that is added to our links
> script.
> 
> BUG=395700
> TESTED=Ran one of our tests with:
> ninja -C out/Release tools_unittests_run
> tools/isolate_driver.py run --isolated out/Release/tools_unittests.isolated --isolate webrtc/tools/tools_unittests.isolate
> 
> R=henrika@webrtc.org, jam@chromium.org
> 
> Review URL: https://webrtc-codereview.appspot.com/26649004

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7328 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 08:44:00 +00:00
kjellander@webrtc.org
02cd3067d2 Update isolate.gypi files + link to isolate_driver.py 
This updates the isolate.gypi copies we're forced to
maintain in our code repo to Chromium revision c264a05.

Since isolated testing is now using a new launch script
in tools: isolate_driver.py, that is added to our links
script.

BUG=395700
TESTED=Ran one of our tests with:
ninja -C out/Release tools_unittests_run
tools/isolate_driver.py run --isolated out/Release/tools_unittests.isolated --isolate webrtc/tools/tools_unittests.isolate

R=henrika@webrtc.org, jam@chromium.org

Review URL: https://webrtc-codereview.appspot.com/26649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7327 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 08:34:57 +00:00
glaznev@webrtc.org
359d720004 Allow Android apps to set video renderer scaling type. 
Also add type check for EGL context object received from apps and
switch to byte buffer video decoding if EGL context is incorrect

BUG=3851
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7326 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-29 23:07:08 +00:00
jiayl@webrtc.org
7dfb7fa189 Reland disallowing blocking calls on the worker thread. 
This fixed the issue that invoking the call when the thread is not started.

BUG=3559
R=juberti@webrtc.org, thorcarpenter@google.com

Review URL: https://webrtc-codereview.appspot.com/24769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7325 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-29 22:45:55 +00:00
henrike@webrtc.org
ea6c12e59f Set thread scheduling parameters inside the new thread. 
This makes it possible to restrict threads from modifying scheduling
parameters of another thread in the Chrome Linux sandbox.

BUG=
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7324 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-29 18:25:27 +00:00
asapersson@webrtc.org
626624061e Disable flaky tests: 
JsepPeerConnectionP2PTestClient.ReceivedBweStatsCombined
JsepPeerConnectionP2PTestClient.ReceivedBweStatsNotCombined

BUG=3871
R=henrike@webrtc.org, kjellander@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7323 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-29 14:30:07 +00:00
kjellander@webrtc.org
e794c36637 Fix parallel test execution for tools, testsupport and metrics tests. 
BUG=2600
TESTED=Passing tests using:
python third_party/gtest-parallel/gtest-parallel -w 10 -r 20 out/Release/test_support_unittests
python third_party/gtest-parallel/gtest-parallel -w 10 -r 20 out/Release/tools_unittests
python third_party/gtest-parallel/gtest-parallel -w 10 -r 20 out/Release/video_engine_tests

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7322 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-29 11:47:28 +00:00
bjornv@webrtc.org
d71118194f audio_processing: Replaced macro WEBRTC_SPL_LSHIFT_W16 with << 
A trivial macro that serves no purpose. Affected components are:
* audio_processing/nsx
* audio_processing/agc
* audio_processing/aecm
* common_audio/LpcToReflCoef

BUG=3348,3353
TESTED=locally on linux
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7321 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-29 10:56:27 +00:00
bjornv@webrtc.org
7c15510f38 common_audio refactoring: Removed macro WEBRTC_SPL_LSHIFT_U32 
The macro is a trivial shift operator including a cast before shift. There is no guard against negative shifts. Replaced with << at place and added casts when necessary.

Affects both fixed and float point versions of iSAC

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7320 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-29 09:40:38 +00:00
houssainy@google.com
24f62e1a28 Adding getStats function to the exposed PeerConnection in RtcBot 
Exposed Peerconnection object has new function "getStats". This function
returns the stats as array of reports, and each report is RTCStatReport
with additional attributes names and stats.

names: array of all the stat names in current report.
Stats: dictionary and the key is the stat name, and value is the value
of this stat.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7319 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-29 09:36:28 +00:00
pbos@webrtc.org
730d270771 Remove callback from RtpDepacketizer::Parse(). 
BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30489004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7318 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-29 08:00:22 +00:00
kjellander@webrtc.org
f21ea918ad GN: Add common configs to all targets. 
This is needed to ensure we have the same build with GN
as with GYP, since GYP includes the common.gypi on a global level.
Several fixes has been needed in the past because some code have
been built without the right defines.

BUG=3441
R=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/28589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7317 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-28 17:37:22 +00:00
pbos@webrtc.org
34f2a9ea72 Initialize SSL in unittest_main.cc. 
Instead of having each test individually initialize and tear down SSL
move this to unittest_main.cc so that all tests are properly
initialized and new tests "don't have to think about it".

R=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/30549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7316 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-28 11:36:45 +00:00